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RetroArch/audio/utils.c
Themaister 33e3eee5ab Nit.
2014-05-23 09:53:27 +02:00

276 lines
9.0 KiB
C
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/* RetroArch - A frontend for libretro.
* Copyright (C) 2010-2014 - Hans-Kristian Arntzen
* Copyright (C) 2014 - Ali Bouhlel ( aliaspider@gmail.com )
*
* RetroArch is free software: you can redistribute it and/or modify it under the terms
* of the GNU General Public License as published by the Free Software Found-
* ation, either version 3 of the License, or (at your option) any later version.
*
* RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
* PURPOSE. See the GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along with RetroArch.
* If not, see <http://www.gnu.org/licenses/>.
*/
#include "../boolean.h"
#include "utils.h"
#include "../general.h"
#include "../performance.h"
#if defined(__SSE2__)
#include <emmintrin.h>
#elif defined(__ALTIVEC__)
#include <altivec.h>
#endif
void audio_convert_s16_to_float_C(float *out,
const int16_t *in, size_t samples, float gain)
{
size_t i;
gain = gain / 0x8000;
for (i = 0; i < samples; i++)
out[i] = (float)in[i] * gain;
}
void audio_convert_float_to_s16_C(int16_t *out,
const float *in, size_t samples)
{
size_t i;
for (i = 0; i < samples; i++)
{
int32_t val = (int32_t)(in[i] * 0x8000);
out[i] = (val > 0x7FFF) ? 0x7FFF : (val < -0x8000 ? -0x8000 : (int16_t)val);
}
}
#if defined(__SSE2__)
void audio_convert_s16_to_float_SSE2(float *out,
const int16_t *in, size_t samples, float gain)
{
float fgain = gain / UINT32_C(0x80000000);
__m128 factor = _mm_set1_ps(fgain);
size_t i;
for (i = 0; i + 8 <= samples; i += 8, in += 8, out += 8)
{
__m128i input = _mm_loadu_si128((const __m128i *)in);
__m128i regs[2] = {
_mm_unpacklo_epi16(_mm_setzero_si128(), input),
_mm_unpackhi_epi16(_mm_setzero_si128(), input),
};
__m128 output[2] = {
_mm_mul_ps(_mm_cvtepi32_ps(regs[0]), factor),
_mm_mul_ps(_mm_cvtepi32_ps(regs[1]), factor),
};
_mm_storeu_ps(out + 0, output[0]);
_mm_storeu_ps(out + 4, output[1]);
}
audio_convert_s16_to_float_C(out, in, samples - i, gain);
}
void audio_convert_float_to_s16_SSE2(int16_t *out,
const float *in, size_t samples)
{
__m128 factor = _mm_set1_ps((float)0x8000);
size_t i;
for (i = 0; i + 8 <= samples; i += 8, in += 8, out += 8)
{
__m128 input[2] = { _mm_loadu_ps(in + 0), _mm_loadu_ps(in + 4) };
__m128 res[2] = { _mm_mul_ps(input[0], factor), _mm_mul_ps(input[1], factor) };
__m128i ints[2] = { _mm_cvtps_epi32(res[0]), _mm_cvtps_epi32(res[1]) };
__m128i packed = _mm_packs_epi32(ints[0], ints[1]);
_mm_storeu_si128((__m128i *)out, packed);
}
audio_convert_float_to_s16_C(out, in, samples - i);
}
#elif defined(__ALTIVEC__)
void audio_convert_s16_to_float_altivec(float *out,
const int16_t *in, size_t samples, float gain)
{
const vector float gain_vec = { gain, gain , gain, gain };
const vector float zero_vec = { 0.0f, 0.0f, 0.0f, 0.0f};
// Unaligned loads/store is a bit expensive, so we optimize for the good path (very likely).
if (((uintptr_t)out & 15) + ((uintptr_t)in & 15) == 0)
{
size_t i;
for (i = 0; i + 8 <= samples; i += 8, in += 8, out += 8)
{
vector signed short input = vec_ld(0, in);
vector signed int hi = vec_unpackh(input);
vector signed int lo = vec_unpackl(input);
vector float out_hi = vec_madd(vec_ctf(hi, 15), gain_vec, zero_vec);
vector float out_lo = vec_madd(vec_ctf(lo, 15), gain_vec, zero_vec);
vec_st(out_hi, 0, out);
vec_st(out_lo, 16, out);
}
audio_convert_s16_to_float_C(out, in, samples - i, gain);
}
else
audio_convert_s16_to_float_C(out, in, samples, gain);
}
void audio_convert_float_to_s16_altivec(int16_t *out,
const float *in, size_t samples)
{
// Unaligned loads/store is a bit expensive, so we optimize for the good path (very likely).
if (((uintptr_t)out & 15) + ((uintptr_t)in & 15) == 0)
{
size_t i;
for (i = 0; i + 8 <= samples; i += 8, in += 8, out += 8)
{
vector float input0 = vec_ld( 0, in);
vector float input1 = vec_ld(16, in);
vector signed int result0 = vec_cts(input0, 15);
vector signed int result1 = vec_cts(input1, 15);
vec_st(vec_packs(result0, result1), 0, out);
}
audio_convert_float_to_s16_C(out, in, samples - i);
}
else
audio_convert_float_to_s16_C(out, in, samples);
}
#elif defined(HAVE_NEON)
void audio_convert_s16_float_asm(float *out, const int16_t *in, size_t samples, const float *gain); // Avoid potential hard-float/soft-float ABI issues.
static void audio_convert_s16_to_float_neon(float *out, const int16_t *in, size_t samples,
float gain)
{
size_t aligned_samples = samples & ~7;
if (aligned_samples)
audio_convert_s16_float_asm(out, in, aligned_samples, &gain);
// Could do all conversion in ASM, but keep it simple for now.
audio_convert_s16_to_float_C(out + aligned_samples, in + aligned_samples,
samples - aligned_samples, gain);
}
void audio_convert_float_s16_asm(int16_t *out, const float *in, size_t samples);
static void audio_convert_float_to_s16_neon(int16_t *out, const float *in, size_t samples)
{
size_t aligned_samples = samples & ~7;
if (aligned_samples)
audio_convert_float_s16_asm(out, in, aligned_samples);
audio_convert_float_to_s16_C(out + aligned_samples, in + aligned_samples,
samples - aligned_samples);
}
#elif defined(_MIPS_ARCH_ALLEGREX)
void audio_convert_s16_to_float_ALLEGREX(float *out,
const int16_t *in, size_t samples, float gain)
{
#ifdef DEBUG
// Make sure the buffer is 16 byte aligned, this should be the default behaviour of malloc in the PSPSDK.
// Only the output buffer can be assumed to be 16-byte aligned.
rarch_assert(((uintptr_t)out & 0xf) == 0);
#endif
size_t i;
gain = gain / 0x8000;
__asm__ (
".set push \n"
".set noreorder \n"
"mtv %0, s200 \n"
".set pop \n"
::"r"(gain));
for (i = 0; i + 16 <= samples; i += 16)
{
__asm__ (
".set push \n"
".set noreorder \n"
"lv.s s100, 0(%0) \n"
"lv.s s101, 4(%0) \n"
"lv.s s110, 8(%0) \n"
"lv.s s111, 12(%0) \n"
"lv.s s120, 16(%0) \n"
"lv.s s121, 20(%0) \n"
"lv.s s130, 24(%0) \n"
"lv.s s131, 28(%0) \n"
"vs2i.p c100, c100 \n"
"vs2i.p c110, c110 \n"
"vs2i.p c120, c120 \n"
"vs2i.p c130, c130 \n"
"vi2f.q c100, c100, 16 \n"
"vi2f.q c110, c110, 16 \n"
"vi2f.q c120, c120, 16 \n"
"vi2f.q c130, c130, 16 \n"
"vmscl.q e100, e100, s200 \n"
"sv.q c100, 0(%1) \n"
"sv.q c110, 16(%1) \n"
"sv.q c120, 32(%1) \n"
"sv.q c130, 48(%1) \n"
".set pop \n"
:: "r"(in + i), "r"(out + i));
}
for (; i < samples; i++)
out[i] = (float)in[i] * gain;
}
void audio_convert_float_to_s16_ALLEGREX(int16_t *out,
const float *in, size_t samples)
{
#ifdef DEBUG
// Make sure the buffers are 16 byte aligned, this should be the default behaviour of malloc in the PSPSDK.
// Both buffers are allocated by RetroArch, so can assume alignment.
rarch_assert(((uintptr_t)in & 0xf) == 0);
rarch_assert(((uintptr_t)out & 0xf) == 0);
#endif
size_t i;
for (i = 0; i + 8 <= samples; i += 8)
{
__asm__ (
".set push \n"
".set noreorder \n"
"lv.q c100, 0(%0) \n"
"lv.q c110, 16(%0) \n"
"vf2in.q c100, c100, 31 \n"
"vf2in.q c110, c110, 31 \n"
"vi2s.q c100, c100 \n"
"vi2s.q c102, c110 \n"
"sv.q c100, 0(%1) \n"
".set pop \n"
:: "r"(in + i), "r"(out + i));
}
for (; i < samples; i++)
{
int32_t val = (int32_t)(in[i] * 0x8000);
out[i] = (val > 0x7FFF) ? 0x7FFF : (val < -0x8000 ? -0x8000 : (int16_t)val);
}
}
#endif
void audio_convert_init_simd(void)
{
#if defined HAVE_NEON
unsigned cpu = rarch_get_cpu_features();
audio_convert_s16_to_float_arm = cpu & RETRO_SIMD_NEON ?
audio_convert_s16_to_float_neon : audio_convert_s16_to_float_C;
audio_convert_float_to_s16_arm = cpu & RETRO_SIMD_NEON ?
audio_convert_float_to_s16_neon : audio_convert_float_to_s16_C;
#endif
}
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